End-to-end delay has a limited time budget in conversational applications, such as telephony. An end-to-end delay exceeding a certain limit will have a negative effect on conversational quality. Therefore it is desirable to keep the end-to-end delay within said certain limit.
The end-to-end delay is the sum of a number of different delays, associated with different parts of the end-to-end communication chain. For example, the end-to-end communication chain typically comprises, besides the actual propagation through a medium, different successive elements, such as source coding, packetizing, channel coding, interleaving, retransmissions and jitter buffering. Each of these elements requires some processing time at their disposal, in order to achieve proper operation of the respective element, such as good source coding efficiency or a reliable transmission over a communication link with only few transmission errors or frame/packet losses. Since end-to-end delay is a restricted resource, there is only a limited amount of time, which can be distributed among the different elements of the end-to-end communication chain.
Before transmitting voice data over a communication link, the voice data is source coded. Typically, source coding is performed in order to compress a sequence of data, in order for it to require, e.g., less transmission bandwidth. Source coding of voice data introduces a source coding delay component in end-to-end delay in conversational services. The length of the source coding delay depends on the type of source codec used, and is typically in the range of 20 ms, as e.g. in GSM EFR (Global System for Mobile Communications) (Enhanced Full Rate), to approximately 50 ms, as e.g. in ITU-T standard G.718 (International Telecommunication Union-Telecommunication Standardization Sector).
One type of source codec, which is frequently used for both circuit switched (CS) and packet switched (ES) transmissions, is the AMR (Adaptive Multi Rate) Multi-mode codec. The AMR Multi-mode codec introduces a coding delay of 25 ms, of which 5 ms is a “lookahead” intended for enabling more efficient source coding. Thus, an AMR Multi-mode codec uses 25 ms of the available end-to-end delay. FIG. 1 illustrates the lookahead in an exemplary AMR codec. Each slot represents 5 ms. In this example of AMR encoder, each frame 106, 108, comprising speech data to be encoded, represents 4×5 ms=20 ms. In the signal analysis of the encoder, a 30 ms windowing frame 102, 104 is used. The windowing frame 104 extends 5 ms into the next frame of speech data, which 5 ms constitute the so-called lookahead 110.
The transmission delay, i.e. the part of an available end-to-end delay spent on the actual transmission through the communication system including, e.g. transmission through the radio access network and transport through the core network, depends largely on the access technology used and on the network configuration. Typically, there is a distinct difference between CS transmissions and PS transmissions, in terms of transmission delay. In CS transmissions, the transmission delay is well defined and fixed for each transported frame throughout the connection. In PS transmissions, on the other hand, each packet is transmitted individually and hence associated with an individual transmission delay, which gives rise to the known packet delay jitter, i.e. variations, in PS transmissions. That is, the packets in PS transmissions arrive at the receiver with different delay, some may arrive early, some late. For CS transmission, access techniques such as 3GPP (3rd Generation Partnership Project) GSM/Geran (GSM EDGE Radio Access Network, where EDGE stands for Enhanced Data rates for GSM Evolution) or UTRAN (UMTS Terrestrial Radio Access Network, where UMTS stands for Universal Mobile Telecommunications System), are used. For PS transmission, e.g. UTRAN PS bearers such as HSPA (High Speed Packet Access) and LTE (Long Term Evolution), are available. VoIP via Internet is also an example of PS transmission.
Even though there are a huge variety of access and transmission techniques available, it could generally be said that extending the available transmission delay budget will make a transmission more reliable, given a defined transmission cost. In other terms, by extending the available transmission delay, the transmission capacity may be significantly increased. For example, in an LTE access network, depending on the operating point in terms of delay and capacity, an increase of the available or allowed transmission delay of as little as 5 ms, may translate to capacity improvements of as much as 15%, regarding the amount of voice calls that can be accommodated in a cell. This is illustrated in FIG. 2, which shows how an increase of the transmission delay of xms in an operating point P1 results in a capacity increase of ΔC1 users per cell. However, the same increase of the transmission delay in another operating point, P2, results in a capacity increase of ΔC2 users per cell, which is considerably smaller than ΔC1. Accordingly, the scarcity of time available for transmission may have a substantial limiting effect on transmission capacity.
The limitation of transmission capacity due to scarcity of available transmission time is thus identified as a problem.